SearcharxivSearch

arXiv · 2606.01009

MelT: A Portable, Single-GEMM Mel Audio Frontend via Non-Uniform DFT with Measured Latency and Energy Gains on GPUs

Abstract

Modern neural audio models run on accelerators whose peak throughput comes from dense matrix multiplication, increasingly at the edge and in datacenters. The conventional acoustic frontend, however -- a Short-Time Fourier Transform (STFT) followed by sparse Mel aggregation -- remains a multi-stage pipeline centered on the Fast Fourier Transform (FFT), with execution overheads unlike the dense linear algebra dominating the inference stack. This work introduces MelT, a portable single-stage Mel frontend that precomputes Mel-spaced Non-Uniform Discrete Fourier Transform (NDFT) bases and applies them to time-domain frames through General Matrix Multiplication (GEMM). The contribution is a computational design principle: decoupling Mel feature extraction from vendor-specific FFT primitives and lowering it onto the matrix-multiplication substrate accelerators already optimize. It is not a new spectral operator. MelT's direct projection performs more arithmetic than the FFT pipeline. Yet in the compact-resolution regime of neural audio frontends, it achieves a 1.64-times to 3.29-times latency reduction and up to a 3.03-times reduction in measured active energy, from the Apple A18 Pro to the NVIDIA H100. All gains are within-platform comparisons, accompanied by task-level validation. Word error rate stays statistically equivalent to the native frontend's on frozen Whisper models of medium size and larger; speaker-attribute classification on VoxCeleb1 is non-inferior. The cepstral extension MFCCT preserves utility on a clinical respiratory-insufficiency classification task (SPIRA) while improving on the MFCC baseline. These results indicate that, in practical regimes on the accelerators evaluated here, hardware alignment rather than arithmetic count can govern the realized cost of feature extraction.

Explore related subjects

Keep this discovery

BibTeXRIS

Augusto Camargo. 2026-05-31. MelT: A Portable, Single-GEMM Mel Audio Frontend via Non-Uniform DFT with Measured Latency and Energy Gains on GPUs. https://arxiv.org/abs/2606.01009

Cite the original work for its findings. Save a collection to share your selection of sources.

KEEP EXPLORING

Related papers

Xiaomi-CocktailASR-1 Technical Report

Recently, large language model (LLM) based ASR models have achieved significant progress, yet they generally lack support for multi-speaker scenarios, where the cocktail party problem remains a critical bottleneck for further advancing ASR. Existing TS-ASR methods, including end-to-end architectures with speaker embeddings and latest LLM-based explorations suffer from degraded single-speaker performance and the inability to reject when the target speaker is absent. In this paper, we propose Xiaomi-CocktailASR-1, an LLM-based end-to-end TS-ASR architecture. By utilizing reference speech as voiceprint prompts, it directly transcribes the target speaker's speech without requiring speech separation. Xiaomi-CocktailASR-1 maintains competitive performance in single-speaker scenarios, comparable to mainstream ASR models. It also features a negative sample rejection capability, outputting empty text when the target speaker is absent from the mixed speech. Additionally, Xiaomi-CocktailASR-1 supports a Chain-of-Thought (CoT) reasoning mode to provide explicit reasoning steps. Extensive experiments on various synthetic and real-world multispeaker benchmarks demonstrate that Xiaomi-CocktailASR-1 achieves state-of-the-art performance, effectively addressing the cocktail party problem through a unified architecture that balances multispeaker and single-speaker recognition accuracy, along with rejection capability.

cs.SD

EConv-TasNet: Efficient Conv-TasNet for Effective Speech Separation

Conv-TasNet has served as a strong baseline for time-domain speech separation, and many studies have extended it with advanced architectures such as dual-path networks, U-Nets, and attention mechanisms. However, these methods often introduce high computational cost and complexity, limiting their deployment in resource-constrained scenarios. To address this issue, we propose eConv-TasNet, an efficient variant of Conv-TasNet that improves both effectiveness and efficiency without relying on resource-intensive modules. The proposed model consists of a group-wise early-splitting (GES) module and a multi-group feature aggregation (MGFA) module. GES generates discriminative speaker embeddings at intermediate stages, while MGFA progressively aggregates these group-level representations for refined mask estimation. Experimental results show that eConv-TasNet reduces model size by 22.4%, accelerates inference by 18.9%, and improves SI-SNRi by 14.0%-28.0% across three public benchmarks. Moreover, it achieves competitive performance compared with state-of-the-art methods while requiring significantly fewer parameters and lower inference cost. These results demonstrate a favorable efficiency-effectiveness trade-off for edge deployment.

cs.SD